Livestream: add silent audio track if not audio input is provided (fixes youtube and other sites)
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dec05eba
parent
c354d88583
commit
e80d7676d4
@ -1,7 +1,7 @@
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[package]
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name = "gpu-screen-recorder"
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type = "executable"
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version = "1.1.0"
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version = "1.2.0"
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platforms = ["posix"]
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[config]
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45
src/main.cpp
45
src/main.cpp
@ -645,7 +645,7 @@ static AVCodecContext *create_video_codec_context(AVFormatContext *av_format_con
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codec_context->codec_tag = MKTAG('h', 'v', 'c', '1');
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switch(video_quality) {
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case VideoQuality::MEDIUM:
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codec_context->bit_rate = 5000000 + (codec_context->width * codec_context->height) / 2;
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codec_context->bit_rate = 6000000 + (codec_context->width * codec_context->height) / 2;
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/*
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if(use_hevc) {
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codec_context->qmin = 20;
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@ -796,7 +796,7 @@ static void open_video(AVCodecContext *codec_context,
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AVDictionary *options = nullptr;
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switch(video_quality) {
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case VideoQuality::MEDIUM:
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av_dict_set_int(&options, "qp", 36, 0);
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av_dict_set_int(&options, "qp", 35, 0);
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//av_dict_set(&options, "preset", "hq", 0);
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break;
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case VideoQuality::HIGH:
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@ -815,7 +815,7 @@ static void open_video(AVCodecContext *codec_context,
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if(is_livestream) {
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av_dict_set_int(&options, "zerolatency", 1, 0);
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av_dict_set(&options, "preset", "llhq", 0);
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//av_dict_set(&options, "preset", "llhq", 0);
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}
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av_opt_set(&options, "rc", "vbr", 0);
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@ -1432,6 +1432,12 @@ int main(int argc, char **argv) {
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const AVOutputFormat *output_format = av_format_context->oformat;
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const bool is_livestream = is_livestream_path(filename);
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// (Some?) livestreaming services require at least one audio track to work.
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// If not audio is provided then create one silent audio track.
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if(is_livestream && requested_audio_inputs.empty()) {
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fprintf(stderr, "Info: live streaming but no audio track was added. Adding a silent audio track\n");
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requested_audio_inputs.push_back({ "", "gsr-silent" });
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}
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//bool use_hevc = strcmp(window_str, "screen") == 0 || strcmp(window_str, "screen-direct") == 0;
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if(video_codec != VideoCodec::H264 && strcmp(container_format, "flv") == 0) {
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@ -1472,9 +1478,14 @@ int main(int argc, char **argv) {
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const int num_channels = audio_codec_context->ch_layout.nb_channels;
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#endif
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if(sound_device_get_by_name(&audio_tracks.back().sound_device, audio_input.name.c_str(), audio_input.description.c_str(), num_channels, audio_codec_context->frame_size) != 0) {
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fprintf(stderr, "failed to get 'pulse' sound device\n");
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exit(1);
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if(audio_input.name.empty()) {
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audio_tracks.back().sound_device.handle = NULL;
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audio_tracks.back().sound_device.frames = 0;
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} else {
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if(sound_device_get_by_name(&audio_tracks.back().sound_device, audio_input.name.c_str(), audio_input.description.c_str(), num_channels, audio_codec_context->frame_size) != 0) {
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fprintf(stderr, "failed to get 'pulse' sound device\n");
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exit(1);
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}
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}
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++audio_stream_index;
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@ -1609,10 +1620,13 @@ int main(int argc, char **argv) {
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int64_t pts = 0;
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const double target_audio_hz = 1.0 / (double)audio_track.codec_context->sample_rate;
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double received_audio_time = clock_get_monotonic_seconds();
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const int64_t timeout_ms = std::round((1000.0 / (double)audio_track.codec_context->sample_rate) * 1000.0);
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while(running) {
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void *sound_buffer;
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int sound_buffer_size = sound_device_read_next_chunk(&audio_track.sound_device, &sound_buffer);
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int sound_buffer_size = -1;
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if(audio_track.sound_device.handle)
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sound_buffer_size = sound_device_read_next_chunk(&audio_track.sound_device, &sound_buffer);
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const bool got_audio_data = sound_buffer_size >= 0;
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const double this_audio_frame_time = clock_get_monotonic_seconds();
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@ -1647,6 +1661,23 @@ int main(int argc, char **argv) {
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}
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}
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if(!audio_track.sound_device.handle) {
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// TODO:
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//audio_track.frame->data[0] = empty_audio;
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received_audio_time = this_audio_frame_time;
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swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&empty_audio, audio_track.codec_context->frame_size);
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audio_track.frame->pts = pts;
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pts += audio_track.frame->nb_samples;
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ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
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if(ret >= 0){
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receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_track.frame, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, *write_output_mutex);
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} else {
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fprintf(stderr, "Failed to encode audio!\n");
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}
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usleep(timeout_ms * 1000);
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}
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if(got_audio_data) {
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// TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?
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swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&sound_buffer, audio_track.sound_device.frames);
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@ -283,7 +283,9 @@ int sound_device_get_by_name(SoundDevice *device, const char *device_name, const
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}
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void sound_device_close(SoundDevice *device) {
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pa_sound_device_free((pa_handle*)device->handle);
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if(device->handle)
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pa_sound_device_free((pa_handle*)device->handle);
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device->handle = NULL;
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}
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int sound_device_read_next_chunk(SoundDevice *device, void **buffer) {
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